This invention relates to mixing audio signals, and more particularly to mixing multiple digital audio signals in a multi-party teleconference.
In the continuing search for more reliable audio transmission systems, many different approaches have been employed. One technique that has recently come of age for practical applications, involves conversion of the analog audio (i.e "voice") signal to "sampled and quantized" digital data signals, which are also known as pulse code modulated ("PCM") signals. With this technique, the analog signal is sampled at regular time intervals. Each sample thus acquired is then assigned a digitally-encoded number (e.g., between -128 and +127) which most closely matches the corresponding analog signal level within an assigned range. Thus, for example, an analog voltage range of -1.28 to +1.27 volts can be represented by an 8-bit binary word (having 2.sup.8 =256 possible values) with a resolution of 0.01 volts so that at near-peak levels the digitally-sampled value is within less than one percent of the actual original analog level sampled. Alternatively, for very fine resolution, 16-bit binary words can be used to accomplish 65,536 quantization levels, thus yielding superb accuracy.
By sampling at a rate of at least twice the "highest frequency of interest" of the audio signal (i.e. at or above the "Nyquist rate") the sampled data can later be reconverted to audio with negligible loss of sound quality. The traditional bandwidth used, for example, by telephone companies for transmitting highly "recognizable" speech was 300 to 3000 Hertz. Thus by sampling at a rate of 8000 samples per second (i.e. at more than twice the traditional 3000 Hertz upper frequency level), excellent quality speech transmission can be accomplished.
The major advantage of this type of digital audio transmission is its "repeatability" with little or no noise increase. Thus, for example digital bits can be received, "cleaned up," and retransmitted with the newly-transmitted signal having no difference from the original signal other than a slight time delay. This cannot be accomplished with analog audio systems.
The potential downside to digital audio transmission is that a much broader frequency bandwidth transmission channel is needed to accommodate the digital audio signals than would otherwise be necessary for the corresponding analog audio signals. With the recent advent of very broad bandwidth channels (especially, example, with fibre-optics cables) digital audio transmission has become increasingly practically and economically realizable.
Computer-based teleconferencing, employing personal computers ("PCs") for example, is a natural application of digital audio since the video portions of the transmissions are all digital. Thus the digital transmission systems used for the video data can also be used for the audio signals.
It is important that the audio portion of a teleconference involving several parties be transmitted in a manner that allows each party to the conference to receive the voices of the other conference members only. That is, it is important that the audio signal received by a participant not include that participant's own voice signal so that an otherwise unacceptably annoying delayed echo received at that participant's station is avoided.
It is also highly desirable for each participant to be able to independently control the volume of the signals received from each of the other participants.